For example, such a method finds technical application in hands-free speaking systems of telecommunications terminals, in which the acoustic coupling between loudspeaker and microphone leads to a part of the incoming signal feeding back through the air path and possibly through a cabinet into the microphone, and thereby to the speaker on the other side of the transmission system. This part is perceived as a disturbing echo by the speaker. The magnitude of the unwanted coupling between loudspeaker and microphone is determined by the type of sound transducers, their distance from each other, their directional effect and sensitivity, and by the ambient conditions under which the sound transducers operate.
It is known to simulate the echo with digital filters, and to subtract it from the echo-affected microphone signal, see Hansler, E.: The Hands-Free Telephone Problem--An Annotated Bibliography. Signal Processing, Volume 27, No. 3, June 1992, pages 259-271. Different methods can be used to determine the filter coefficients of the digital filter, for example a nonlinear canonical algorithm provided with a sign place, see EP 0,310,055 B1, or the Normalized Least Mean Square Algorithm, NLMS-algorithm for short, see T. Huhn, H. J. Jentschel: Combination of Noise Reduction and Echo Cancellation During Hands-free Operation; Electronic Communications Technology, Berlin 43 (1993), pages 274-280.
Several thousand filter coefficients may be required to obtain a useable approximation to the time function of the echoes. A floating point signal processor with a high operating speed and a large memory are required for the practical realization, so that these possibilities can only be attained at high cost. Because of their high cost, the solutions known so far are not suitable, particularly to fulfill the commercial requirements of qualitatively satisfying hands-free speaking systems, for example narrow band telephones, narrow and broad-band video telephones, in video conference studios, public address systems and listening devices.
The difficulty of balancing the filter coefficients lies in precisely measuring the current pulse response on the basis of the acoustic feedback from the loudspeaker via the air path, and partly through solid bodies, for example a cabinet, to the microphone, even in the presence of disturbing influences. Thus when using an echo canceller, the requirement exists to simultaneously and very reliably differentiate:
a) between echoes and local noise, particularly duplex communication, the so-called double-talk situation,
b) between echoes from far reflectors and echoes from near reflectors, and
c) between the kinds of incoming signals that permit taking sufficiently precise measurements, for example sufficiently strong broad-band signals, such as for example noises, as they ideally occur with a pulsed impact, and incoming signals that are not suitable for measurements, such as for example weak signals or a continuous sinusoidal signal.
To partially fulfill these requirements it was already proposed to calculate the filter coefficients of a Finite Impulse Response filter, FIR for short, in accordance with the known NLMS algorithm, and when calculating a new filter coefficient, to make the responsible step width .alpha. switchable as a function of the magnitude of the potential echo signal, see U.S. patent application Ser. No. 08/197,054, filed Feb. 16, 1994 (U.S. Pat. No. 5,467,394), claiming priority from German Patent Application P 43 05 256.8, filed Feb. 20, 1993. To preclude improper adjustments during duplex communication, the step width .alpha. is set to zero in this case, so that a change in the filter coefficients is avoided. However, the filter coefficients are also calculated when the local speaker is silent, and this also means that the filter coefficients are calculated anew, even in the presence of very weak input signals, for example noise, thereby resulting in an improper adjustment of the echo canceller. It was also proposed in the last-mentioned application to improve the echo suppression by determining a loudspeaker-microphone coupling factor, and to vary the loudness of the loudspeaker or the sensitivity of the microphone as a function thereof. The loudspeaker-microphone coupling factor is determined by the correlation of the microphone signal with the loudspeaker signal. However, such a correlation analysis has the disadvantage that erroneous decisions of the correlator cannot be excluded because of the similarity of the speech signals. In addition, the calculation effort is also very extensive, so that the correlation analysis alone requires about 30% of the entire program cost.